93results about "Network signaling arrangements" patented technology

A Voice over Internet Protocol (VoIP) network is described in which session state is maintained in access switches, but not signaling gateways which maintain transaction state only during pendency of a related transaction. The signaling gateway further provides transparent inter-operation between the VoIP network and non-IP networks, such as the PSTN, by means of a translator which directly translates messages between the networks.

The present invention provides a system and method for a reliable, low-cost, secure Internet Protocol (IP) based network that provides broadband-based voice communications as well as video and data communications. The IP network is arranged to function with the infrastructure of the Public Switched Telephone Network (PSTN), to control telephone calls in SS7 type networks and to provide the features, applications, and services of the typical SS7 networks in a voice over IP network. The present invention supports large effective call volumes, allows accommodation of a wide range of broadband-based service platforms, provides flexibility to support current and future calling feature services, and provides high quality voice transmission.

A method of setting up a session between peer user terminals of a communication system, said session extending at least in part across a circuit switched access network. The method comprises transporting signalling to initiate said session between at least one of the peer user terminals and said communication system via an IP based packet switched access network using a call control protocol which is also used for setting up end-to-end packet switched sessions, and subsequently establishing said session based upon said signalling.

Call control and related messaging may be communicated between systems that operate using different protocols. In one version, a process is described that includes generating a message in a first telephony signaling protocol, wrapping the message in a message of a second telephony signaling protocol, and sending the wrapped message to a telephony switch using the second telephony signaling protocol.

A SIP-based network architecture is defined allowing a variety of access methods for interfacing QSIG-based signaling devices. The QSIG-based signaling devices are typically enterprise-based PBXs that either interface directly via QSIG, or adapt QSIG signaling to related signaling protocols via gateways, which can be customer premise or network based. The gateways can either interwork QSIG signaling or tunnel QSIG signaling over H.323, SIP, or other protocols. Upon reaching the SIP-based network, Session Border Controllers process the signaling. The Session Border Controllers terminate the tunnel (if used) and extracts the QSIG signaling for interworking to SIP, thereby facilitating a homogeneous SIP-based core network. In addition to interworking QSIG signaling, the Session Border Controllers perform other signaling related functions for the SIP-based network that are performed at a point as near as possible to the network's edge.

A subscriber's data circuit between channel bank and the customer premises carries fractional T1 bandwidth on a digital subscriber line (DSL) circuit. For example, for an ISDN rate digital subscriber line (IDSL), the circuit between the network termination at the customer premises and a D4 channel bank carries data on two combined B-channels. Such a circuit also carries a D-channel and an embedded operations channel (EOC), in normal ISDN fashion. A data service using such a line circuit, however, only transports the data (combined B-channels) through the network. The D-channel and the EOC are not carried through the network. To facilitate testing of the subscriber's circuit, the carrier operations and testing facilities transmit loop-back commands or the like in band to the channel bank. A command may be addressed to any active node along the subscriber's DSL circuit. The line card serving the subscriber in the channel bank detects the commands, removes the commands from the in-band data and reformats the commands for transmission over the EOC channel. The line card addresses the commands and sends the EOC format commands over the DSL circuit to the appropriate node along the subscriber's line. In response, the node connects the circuit in a loop-back mode, to enable test signal transmission and analysis.

A signaling gateway allows a next generation of service nodes to be deployed in any communications switch network without requiring any further development or customization. The signaling gateway encapsulates multiple signaling systems into a single interface embodied in a telecommunication service provider's proprietary signaling protocol. The signaling gateway also performs call and resource management. A next generation of service nodes may then be designed to process a reduced set number of messages that have a common format and that perform fundamental functions common to all signaling systems. Any differences in implementation of any signaling system, as well as detailed functions performed for call setup and resource management, are transparent to the service node.

A system for inter-enterprise telephony communications includes a central brokerage device that can receive a packet encoding a dialed telephone identifier generated by a first telephony device located in the first enterprise network, where the dialed telephone identifier is associated with a second telephony device located in a second enterprise network. The central brokerage device can also identify a network address of the second enterprise network using connectivity information if the dialed telephone identifier matches resolution information. The system also includes the first enterprise network, which can establish an encrypted communication link with the second enterprise network through a packet-based network using the identified network address if the dialed telephone identifier matches the resolution information. The first enterprise network can also establish a second communication link between the first enterprise network and the second enterprise network through a circuit-switched network using the dialed telephone identifier if the dialed telephone identifier does not match the resolution information.

A system is provided in which a client/server/network can implement a key management session when the server initiates the key management session utilizing a nonce. The nonce allows a wakeup or trigger message to be conveyed to the client such that a service attack on the server can be avoided when a false nonce is received by the server with an AP request message. Thus the server can disregard AP request messages that are not accompanied by a nonce stored by the server. The method can be implemented through circuitry, electrical signals and code to accomplish the acts described in the method.

A method for analyzing ISVM signaling messages that includes copying, to a message file, messages being transmitted through a signaling link, extracting ISVM messages from the message file that meet at least one trouble criterion, storing the extracted messages in an error report file, and generating an error report from the error report file. The error report lists the extracted messages. Examples of the trouble criterion include messages referencing a designated directory number, messages reporting an error, messages having a signaling point code and a destination point code that are equal, query messages having no matching responses, and unit data service messages.

An Internet protocol (IP)-private branch exchange (PBX) system and a method for linking the system with a Voice over Internet Protocol (VoIP) terminal. The IP-PBX system and method allow the VoIP terminal to be registered at an extension of the IP-PBX system, the VoIP terminal supporting a protocol, e.g., H.323 or session initiation protocol (SIP), standardized by a standardization organization, e.g., the International Telecommunication Union (ITU), the Internet Engineering Task Force (IETF), or the European Telecommunications Standards Institute (ETSI).

In an audio transmission system, a control section of a communication conference device emits measurement sound waves from a loudspeaker array to a terminal unit and measures the time until a response is received, thereby detecting the position of the terminal unit. The control section sets directivity characteristics so that microphone sensitivity of the microphone array is brought to point to the position of the terminal unit, and sends the collected audio to another communication conference device. In a communication conference device on the reception side, the directivity characteristic of the loudspeaker array is set so that the received audio appears as if it was emitted from the position of the terminal unit on the transmission side.

A method, system and gateway device enables interworking between an IP-based network and a circuit-switched network. A first address information of a first connection end located in the circuit-switched network is routed in a trigger message from the IP-based network to a gateway control function. The first and second call legs are established in parallel towards the first connection end based on the first address information, and towards a second connection end located in the IP-based network based on a second address information obtained from the trigger message. A single connection between the two connection ends is then established by connecting the first and second call legs. Thereby, IP-based signaling functionality can be used to add capability for subscribers located in the CS domain to be invited into conferences or calls with subscribers located in an IP-based domain, e.g. the IMS domain.

A system and method to provide secure access across the untrusted public switched telephone network is described. The system and method can be initiated by a security policy defining actions to be taken based upon at least one attribute of the call.

A method of controlling a graphical user interface (GUI) at a wireless device is disclosed and includes storing a set of audio GUI controls at an interactive voice response server and creating an audio GUI control string that is to be communicated to the wireless device within a voice stream. The audio GUI control string corresponds to a text string that is selectably presentable at the wireless device. Further, the method can include embedding the audio GUI control string within the voice stream. Additionally, the method can include transmitting the voice stream with the embedded audio GUI control string to the wireless device.

Users, who have simultaneous access to a plurality of terminal devices, can enter into a multimedia, multiterminal user session. In a negotiation phase users agree on which media to use in a multimedia session and select the most appropriate types of terminal devices that support the selected media. Each terminal device is connected to one of a plurality of interworking communications networks. A network node (130) that is reached through any of the plurality of interworking networks manages creation and supervision of sub-sessions involving the selected terminal devices such that each type of terminal device carry at least one selected media. A versatile and media rich communication is obtained in which terminal devices currently available at a local environment are efficiently used.

If a communication client disconnects from a call, an interworking service node may receive from the communication client a disconnect message indicative of an attempt to disconnect the call and send a forced hold message to place the call on forced hold in response to receiving the disconnect message. A portion of the call extending toward a public service access point and through the PSTN is placed on hold. When an operator wants to reestablish the call to the communication client, the interworking service node will receive over the PSTN a reconnect request message to reestablish the call with the communication client. The reconnect request message may be an integrated services user part message. In response to the reconnect message, the interworking service node will generate and send a session initiation message, which is configured to reestablish the call toward the communication client.

A call control server establishes a connection between a first terminal agent and a second terminal agent, as an existing call. When a first real terminal device included in a first terminal group has accepted a transfer request, the call control server transmits existing-call terminal & media notification data to a second real terminal device included in a second terminal group. The second real terminal device clearly shows a user at the transfer destination the difference between media available in the second terminal group and the media involved in the existing call, and accepts from the user a designation of media to be employed in a transferred call. The call control server establishes a connection between a third real terminal device included in a third terminal group and a real terminal device included in the second terminal group, as the transferred call, in accordance with the designated media.

A communication system comprises a communication platform and a wireless service node. The communication platform receives first signaling for a call to a telephone number, and in response, transfers second signaling to alert a packet network appliance of the call, and the communication platform transfers third signaling. The wireless service node receives the third signaling, and in response, transfers fourth signaling to alert a wireless telephone of the call. The wireless telephone and the packet network appliance are simultaneously alerted of the call.

A method is specified for the handling of a telecommunications connection between at least two subscribers, wherein the telecommunications connection has a signalling Channel for transmission of a signalling stream and a payload Channel for the transmission of a payload stream, wherein frequency-coded Signals are transmitted in the payload stream, wherein at least one subscriber among the at least two subscribers is assigned to a network which has a switching device and a signal processing device provided separately from the switching device, and wherein the frequency-coded Signals are recognised in the signal processing device and are transmitted independently of the payload stream to the switching device. The invention also relates to a telecommunications arrangement, a switching device and a network coupling device.

A method and system for graphically displaying call signaling flows for a communications network is described. In one example, a call flow alarm indicating an improper call signaling flow for a call is received. A respective projected call signaling flow associated with the call is then obtained. Lastly, the improper call signaling flow and the respective projected call signaling flow are graphically displayed.

At least one switching instance is provided in a VoIP system in order to set up a useful data link between a calling terminal and a called terminal. A first signaling link is set up between each terminal and the at least one switching instance for interchanging information on the basis of a first protocol, a second signaling link is set up between them for interchanging information on the basis of a standardization second protocol, in which the useful data link is set up on the basis of the information interchanged on the basis of the second protocol. At least the second signaling link is initiated by the calling terminal, with the information which is transmitted via the first signaling link and in parallel via the second signaling link from the terminals being in each case combined.